RTC_CHECK and RTC_DCHECK macros for C

webrtc/modules/audio_coding/codecs/opus/opus_interface.c

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
 
-#include <assert.h>
 #include <stdlib.h>
 #include <string.h>
 
 enum {
   /* Maximum supported frame size in WebRTC is 60 ms. */
   kWebRtcOpusMaxEncodeFrameSizeMs = 60,
 
   /* The format allows up to 120 ms frames. Since we don't control the other
    * side, we must allow for packets of that size. NetEq is currently limited
    * to 60 ms on the receive side. */
@@ skipping 19 matching lines @@
       opus_app = OPUS_APPLICATION_VOIP;
       break;
     case 1:
       opus_app = OPUS_APPLICATION_AUDIO;
       break;
     default:
       return -1;
   }
 
   OpusEncInst* state = calloc(1, sizeof(OpusEncInst));
-  assert(state);
+  RTC_DCHECK(state);
 
   int error;
   state->encoder = opus_encoder_create(48000, (int)channels, opus_app,
                                        &error);
   if (error != OPUS_OK || !state->encoder) {
     WebRtcOpus_EncoderFree(state);
     return -1;
   }
 
   state->in_dtx_mode = 0;
@@ skipping 389 matching lines @@
     return 0;
   }
 
   for (n = 0; n < channels; n++) {
     if (frame_data[0][0] & (0x80 >> ((n + 1) * (frames + 1) - 1)))
       return 1;
   }
 
   return 0;
 }